On Thu, Oct 6, 2022 at 10:17 AM Jerry Geis <jerry.g...@gmail.com> wrote:
> > > On Thu, Oct 6, 2022 at 9:02 AM Jerry Geis <jerry.g...@gmail.com> wrote: > >> >> >> On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis <jerry.g...@gmail.com> wrote: >> >>> I am trying to get audio to work on AWS using asterisk 18.14.0 >>> >>> I have enabled the firewall to allow ALL UDP on AWS >>> >>> My SIP extension has >>> nat=force_rport,comedia >>> qualify=yes >>> allow=ulaw >>> allow=alaw >>> allow=gsm >>> canreinvite=yes >>> >>> I enable "rtp set debug on" and the console is printing info. >>> >>> The call comes into my linphone softphone - but I get no audio on my >>> linphone softphone. >>> What might I be missing to allow the audio ? >>> Volume is up. >>> >>> Thanks >>> >>> Jerry >>> >> >> >> I just noticed the RTP log is sending to 192.168.2.0 which is my local >> lan address of the linphone - it should be sending to the NAT address and >> is not. >> What did I not set correctly ? >> I am not using pjsip - but the older asterisk. >> >> Thanks >> >> Jerry >> > > >Have you configured chan_sip to know it is behind NAT itself and what its > >public IP address is? If not, then you'll get no audio. > > I'm thinking I have not. What did I miss ? > The sample configuration file outlines how things work, and the options for it: https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874 in general localnet and externip (or externaddr, or externhost) -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org
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